nik600 wrote:
Hi to all
in a scenario where:
- the bandwith is shared with other traffic (HTTP,VPN,ecc)
- the PBX is on a remote VPN peer
- due to many reasons Qos is not usable
There is a IAX trunk between 2 Asterisk 1.4 i've tried different
codecs (ulaw,alaw,gsm) but the main problem
Jason Aarons (US) wrote:
My understanding is Skype's secret is using the iLBC codec, which Cisco
has also licensed for their 79X2 models as well. I travel and lot and
in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator
will fail the Skype client will work. The iLBC codec
Klaverstyn, David C wrote:
Hi All,
I was under the impression that I found a WEB site about two years or so
ago that allowed Asterisk users to place free calls between each other
that used up users un-used minutes/calls. I though the site was IAXtel
but that does not seem to be the
Chad Whitten wrote:
I have been searching for some documentation that would indicate if
Asterisk supports H.248 and everything I have come across seems to
indicate I should use MGCP which I would agree is a better choice but
unfortunately the equipment I am trying to integrate only does H.248.
remember).
You may also want to check if your gateway can't be changed to use
another protocol, since H.248, AFAIK, is not exactly much supported in
the OSS world
On Feb 16, 2008 5:05 PM, Julio Arruda [EMAIL PROTECTED] wrote:
Chad Whitten wrote:
I have been searching for some documentation
Al lists wrote:
Theoretically, setting TOS value ( these days called DSCP) wont change
anything in switch behavior, unless you are using Layer 3 switches.
What makes a difference in a switch is COS bits, and i'm not sure how
asterisk sets that.
I guess to be safe, you would need to create 2
Olivier wrote:
Hi,
1. Is your WiFi network dedicated to VoIP or shared with data applications ?
How was it designed ?
For people using WiFi with a laptop, you propably don't need to have dense
WiFi cells as moving from one cell should be scarce.
With hand phones, those cells should overlap
Fernando Berretta wrote:
Dear Mindaugas,
Thanks for your promt response
I've already tried this but.. it's not working,, what file do you think
I should use ? any other idea ?
Fernando,
I've used the official/legal G729 codec sold at www.digium.com in Athlon
boxes w/ asterisk 1.4
How many licenses you have (show g729 should give you this info)
Scott Moseman wrote:
Gateway sends Asterisk an INVITE (using g729)
Asterisk sends Phone an INVITE (using g711 or g729)
Phone sends Asterisk an OK (using g711)
Asterisk sends Gateway an OK (with no RTP choice)
Gateways ends the
Tzafrir Cohen wrote:
On Fri, Oct 05, 2007 at 08:12:34AM -0500, Brian West wrote:
You can hear and understand someone much better with g722... more
emotion is transfered over the phone when using g722.
G722 is free and in the clear. G722.1 and G722.2 are not.
But speex *Is* free.
Is this a SIP connection or a SIP-T one? Not sure (don't have access to
my previous life docs :-), but this seems to be a Session Server Trunks
doing SIP-T, not sure is the configuration you want...Have you tried to
contact their support ?
PS: this c:
it to regular SIP servers
thusfar though. Also like I mentioned, I don't have this one-way RTP
problem with an earlier version of Asterisk.
Thanks for your reply,
Örn
On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote:
Is this a SIP connection or a SIP-T one? Not sure (don't have access to
my
Just one question, why would the Asterisk be involved in the voice path
at all ?
I would assume a media gateway (TNT ?) would be the obvious choice to
provide trunking side. And, for line side another gateway (not so sure
would be as often seen), but in this case a Line side gateway, and
Gang Chen wrote:
- Original Message -
From: Kristian Kielhofner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 22, 2007 4:32 PM
Subject: Re: [asterisk-users] inband DTMF for g729
On 6/22/07, Gary
Richard Klingler wrote:
Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;
I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...
I've a 7912G
Benjamin Jacob wrote:
rfc2833 is the prefered way, as inband will work perfectly only with the
ulaw codec.
Out of curiosity, there is any 'document' about how RFC2833 would be
better or worse than SIP INFO ?
Pierre Marceau wrote:
Okay, in the SPA-941 admin I changed:
;DTMF Tx
Gordon Henderson wrote:
On Thu, 25 Jan 2007, Yuan LIU wrote:
Thanks for this information. Does this mean two IAX boxes can talk
behind their respective NAT's (without any server sitting in voice
path)? I'm imagining this:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
Using
Doug,
You are saying that RFC2833 somehow doesn't work if you have the
Asterisk AND at a distinct time (still within the same call), the callee
to see the DTMF, correct ? Would this be in any case ? (meaning, if the
voice path is going via the Asterisk or UA to UA directly ?)
I've my spa3k
Doug Crompton wrote:
I am not sure what you are asking? The problem is that rfc2833 does not
play well with the spa-3000 and Asterisk. I am not sure if it is limited
to just the spa3k. There is a bug causing this that has been documented.
Google spa3000 dtmf bug asterisk for more info. The
Eric ManxPower Wieling wrote:
Doug Crompton wrote:
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for such things
SIP over TCP != RTP over TCP
The whole latency deal is much more of a concern in RTP (as well as
trying to deliver a late packet, that will be not very useful also).
As I understand, MS does SIP/TCP on their LCS or something like that.
Still, not RTP over TCP, since it does not make sense for
Darrick Hartman wrote:
Kenneth Padgett wrote:
I'm looking for opinions on the best value router to use for home
offices.
It should work for a scenario in which there are 3 computers and 2 SIP
phones, handling QoS so that the phones always have higher priority
traffic
than the PCs. (and not
Try to search for the PrivacyManager application.
It does 'check' if the CallerID is present, if not, it will play an
announcement to ask the person to 'type' their phone number, and it will
allow you to then accept it.
je . wrote:
Is it possible to reject all incoming calls that do
Benny Amorsen wrote:
MG == Michael Graves [EMAIL PROTECTED] writes:
MG Who will benefit as long as calls must typically pass into
MG existing PSTN infrstructure, and so be transcoded into G.711? It
MG seems to me that only systems that are IP end-to-end stand to show
MG the improvements...or
Eric ManxPower Wieling wrote:
Vikki wrote:
I think vonage is using g723.1 which requires 6.4kbps voice bandwidth
compared to g711 - 64kbps.
For SIP to SIP calls, RTP doesn't necessarily goes thru the server. Only
Signalling goes to the servers. This means no bandwidht usage for the
provider.
Steven wrote:
Because the Telco is government owned.
They are the PSTN, so only they can route and charge for PSTN calls.
Making a call from an Indian office to a US office over VOIP is legal.
Forwarding a PSTN call over that same VOIP trunk is illegal.
In other countries where the Telco is
Dovid Bender wrote:
- Original Message - From: Steve Davies [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 31, 2006 6:01 AM
Subject: Re: [asterisk-users] SNOM 360
On 7/31/06, Koopmann, Jan-Peter
Matt Florell wrote:
Yes, that is very confusing :)
Is there no way to throw a timer chip in there(I suppose it's way too
late to put that suggestion forward now)?
Curiosity, isn't the timer from the 2.6 kernel 'good enough' for
Asterisk purposes nowadays ?
Or there is a constraint using
Abdul Lateef wrote:
Hi everyone,
I was trying to support SRTP in asterisk for our
Linksys IP Phones to prevent of ISP blocking issue.
I compiled successfully SRTP from
http://srtp.sourceforge.net/srtp.html
But i don't know from where i should start to
configure in Asterisk.
Could someone
Bill Gibbs wrote:
Goal – to get the CoralIP PBX long distance savings by sending it to
Asterisk (which then talks via SIP to other long distance voip providers)
The Coral IP supports MGCP and so does Asterisk. Has anyone tried
sending calls from the Coral PBX to Asterisk via MGCP? I will be
Most likely, he is thinking something like using the MTA (a motorola
cable modem with RJ11 phone ports), to register to Asterisk.
From what I understand, most (if not all) packet cable VOIP is done
using NCS (a mgcp-like protocol ?) as call control, not SIP.
Alexander Lopez wrote:
Isn’t
Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ?
From a quick test (got mine yesterday), seems like it is not
recognizing Caller ID from PSTN/FXO port..
Using the same configuration as a Sipura 3000 (to be sent to
mother-in-law POP :-), no Caller ID at all, (I've even
Peter Bowyer wrote:
On 22/05/06, Steve Kennedy [EMAIL PROTECTED] wrote:
On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:
If you want to roam between GSM and WiFi while on a call, the GSM
carrier is going to have to support it.
There is a protocol for this (UMA),
There is a mailing list (in portuguese, but most persons there will
answer your questions in english without problems) in Brazil.
IF your question is biz related, of course, there is a proper place for
these (a biz list).
http://listas.asteriskbrasil.org/mailman/listinfo would give you both
Douglas Garstang wrote:
Does anyone know if it's possible to set the codecs for a number via an
Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a
command that can set the same thing so that it can be done without having to
change sip.conf.
Paulo Scardine wrote:
I have a worst issue for you... If your fax solution is ever going to
receive fax in Brazil, how would you block collect calls?
I have made a fax server solution with cheap Digium hardware that works
in Brazil (2 E1s).
--
Paulo,
He is mentioning E1/PRI, so I assume the
Eric ManxPower Wieling wrote:
Use a codec your phone supports like ulaw.
Assuming he is using SJphone, that I understand, would support iLBC even
in the free version ?
Alyed Tzompa wrote:
made the changes in sip.conf so now it reads:
disallow=all
allow ilbc
now I when the call is
From what I can see
The 2 legs of the call are: 'phone' in alaw and 'laptop' in g726, why
should he need G.729 anywhere ?
Bartosz, not exactly that familiar, but I guess you could try to debug
the call establishmment.
(one thing that puzzles me, you mention IAXy, but you show 2 sip.conf
Since the last hurricane (that left me without phone for around 3 weeks
or so), I did the call forwarding (remote call forwarding in fact).
Lucky I was running in the cable modem in a couple of days (power restored).
I was planning in having two DIDs in distinct providers (I've been using
Rich Adamson wrote:
..
A fairly common assumption is the failover happens in xxx milliseconds, but
due to nic card design (etc) a different MAC address is used in the failover
condition. That confuses the hell out of the layer-3 boxes and negates the
value of the failover. (All documentation,
Rich Adamson wrote:
.
Last, the bonding of two nics at the server level _requires_ the associated
switch interface to support the exact same bonding algorithm. Historically,
that has been a problem for many switch vendors.
Not so sure I understand, but if you mean, 'the algorithm to select a
I was testing Broadvoice few weeks before Hurricane Wilma here in FL.
Since then, I had been since the landline (Bellsouth), and I had to
'remote callfwd' the BS # to my broadvoice #.
So, from my impression, is ok for my needs (I got a weird no ringback
problem that I kind of solved with a
Just to clarify this in my head :-)..
So...
They are using E1/R2 (the R2 Digital)in fact, for all the line signaling
(nothing unusual)
The register signaling, that I was under impression would be MF in each
timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMF
in this
Jesus Mogollon wrote:
Hi Steve:
Thanks for your help. I really appreciate it..
My provider is CANTV in Venezuela. There's a venezuelan variant in the code
and I'm using that. Incoming works perfectly, outgoing is not working. I'm
being told that incoming is MFCR2 but outgoing is R2-Digital
DNS caching server running in the same machine ?
Eric ManxPower Wieling wrote:
Um, put in IP addresses instead of hostnames in Asterisk's config files?
Eric Bishop wrote:
I agree about Asterisk being terrible with DNS failure, but how can you
avoid using DNS on *nix system?
On 11/7/05, Eric
Patrick wrote:
On Wed, 2005-09-28 at 23:17 +0800, Steve Underwood wrote:
[snip]
An effective DOS attack on a $300,000 Alpha running NT I used to use was
wiggle the mouse :-) I never really understood how that brought a
multi-CPU machine to a standstill, but it did.
Reminds me of an
Patrick wrote:
On Sat, 2005-10-01 at 08:31 -0400, Julio Arruda wrote:
[snip]
One thing interesting, coming from data background, seeing the
requirements in carrier voice networks. Is a quite distinct ball-game.
Devices that require 'hot-software-upgrades', still not that often seen
in data
Set Wild guess mode on (I'm not familiar with zyxel).:
asterisk-users wrote:
Has anyone experiences this please: -
We were running a number of ZyXEL P662HW-61 routers at our sites and all
traffic was being sent over IP-SEC VPN's between devices.
When we moved to a new architecture, we got
Matt Riddell wrote:
Adam Robins wrote:
Should it be in half duplex or full duplex?
Full.
AFAIK, depends...
If you have your switches doing autonegotiation, you can't disable
autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL
have a duplex mismatch.
This is as
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Monday, August 29, 2005 8:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Matt Riddell wrote:
Adam Robins wrote:
Should it be in half
Remarks inline
Dean Collins wrote:
Packet8 got around this in an interesting waycharge clients $1.50
per month for E911 or have the option of saying no.
Lol, how many people do you think took them up on that offer?
From what I understand, Packet8 had this option for quite some time. I
Half duplex by itself doesn't hurt (depends in number of calls and etc
really, but anyway...)
What is a killer for VOIP is duplex mismatch.
If you have autonegotiation enabled, and your peer (the switch ?) has
autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex
mismatch.
And
Just wondering..
Any experience with the UIP1868 ?
I assume that it can handle a single SIP line (can't seem to find the
manual at their site :-)..
They mention also T38 in their webpage.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Andrew Kohlsmith wrote:
On Monday 25 July 2005 23:26, [EMAIL PROTECTED] wrote:
Highly recommended to everyone to stay away from this issue
I do not have a name for the company right off hand, but they got sued
really bad when they tried 911 via VOIP and the 911 drop kept occurring in
different
Dave Cotton wrote:
On Fri, 2005-07-22 at 15:42 -0500, Eric Wieling aka ManxPower wrote:
Eric Rees wrote:
We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX
sounds better then the land line.
Using
Not that it need any additional 'push' against it, :-)..
My tests with IAX over OPENVPN (on port 443) are acceptable (they do
work just fine) for basic non-user-friendly purposes.
Examples, I get my voice mail at home sometimes via this tunnel (if wife
using primary landline.
I test my
Just remember that TCP will try to retransmit your lost voice packets,
what is not exactly of any use :-).
VPNs with IPSec and others (CIPE and some UDP 'related' vpns) would not
create this extra (and useless) overhead.
I've used IAX over OpenVPN (with SSL as you), and it does work, to some
to remember all TCP fancy stuff)..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Friday, July 22, 2005 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX over HTTP
Just remember
Rob Scott wrote:
For work environments where you only get HTTP or HTTPS access, what is
the feasibility of doing IAX over HTTP?
I have heard of projects such as stunnel.
Has anyone tried something like this?
I did a quick search but didn't come up with much.
I did some tests, with openvpn,
Dhennys,
I would expect that the ISDN collect call would have some kind of
notification about the charge.
In E1/R2, the Telebras standard in fact DOES have this notification
defined, from what I remember, the problem was that many of the CO
switches would not support it, that is why the
Denis Galvão - iSolve wrote:
IAX doesn't use INBAND DTMF.
Denis Galvão.
Denis,
A clarification, I hope, just to make Mark aware of the small difference.
IAX sends DTMF in the signaling 'stream', that happens to follow the
same path as the media.
But, in IAX DTMF is not sent as voice payload
[EMAIL PROTECTED] wrote:
On Thursday 23 June 2005 19:57, Brian West wrote:
With inband its at least not sent in clear text.
It's trivial to pull DTMF out of an inband stream too. Perhaps not AS
trivial
but just the same, you should be using SRTP if you're paranoid about this
kind of thing.
Andrew Kohlsmith wrote:
On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote:
But there are some products that supports DTMF inband on G729. Ok, it will
not work in most cases(like everyone told) but why Asterisk dont support
it? Is this hardcoded, or is possible to try it out?
Asterisk
[EMAIL PROTECTED] wrote:
2- Out-of-band is as safe/unsafe as having the conversation recorded,
including pin, by the hacker, if no encrypted voice path is being used.
as others mentioned, DTMF tones would be very obvious in a trace
(maybe someone may want to post an example).
Watch out:
Roman Zhovtulya wrote:
Dear all,
I've noticed some significant voice quality deterioration when calling US
landline via VoIPjet.com in the last week or so.
Before that the quality was pretty good.
Has anyone else experienced any voice quality problems with voipjet
recently?
I've been using
Anyone tried the Packet8 Videophones ?
I would guess that leadtek is providing the non-branded version now ?
[], O-O
Dean Collins wrote:
I've played with the dlink eyebeam but only for ip to ip calling not
used with asterisk.
It's crap.
-Original Message-
From: [EMAIL PROTECTED]
InternetMarketingMan2001 wrote:
I want to collocate an * box somewhere, where better than where voicepulse
chose to put their servers?
They probably did their homework and selected someplace where good handoff
to the pstn can be found, right/
AFAIK, the voice path doesn't really need to follow the
Asterisk wrote:
Hi all,
Is there a VOIP provider that can deliver local Rio de Janeiro numbers?
I am looking for a normal Rio number for my Asterisk box.
I'm using a RJ DID, right now http://www.libretel.com with IAX DID (they
offer SP also).
Have not tried much on it, noticed DTMF can be a
Doug Lytle wrote:
Grandstream owners,
I just noticed that there is a new firmware release, for those that are
interested:
http://www.grandstream.com/BETATEST/
2 quick notes, a quick test seem to indicate iLBC is broken (didn't try
any troubleshooting).
And, in the release notes, from what I
Matteo Brancaleoni wrote:
yes, some multiplexer allows that, but they're quite expensive
compared to another E1 card for asterisk.
I think you'll need at least 1k $$$ for a such splitter.
Matteo, would you have any reference for this 'mux/splitter' ?
I would guess it need to be smart enough to
Stefan de Konink wrote:
On Wed, 27 Apr 2005, Joseph wrote:
How can proprietary protocol be open protocol?
If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Digium,
Tom Ivar Helbekkmo wrote:
Leo Ann Boon [EMAIL PROTECTED] writes:
My prediction: 2 years down the road, they'll leave again and set up
SipZilla to make another low-cost ATA to compete with the SPA-.
...and then they'll sell *that* to Cisco, too. :-)
Or, 2 years down the road, VOIP will be
Vikram Rangnekar wrote:
Does anyone know the exact VOIP regulations in India. What I want to know is
that are VOIP EPBX with E1 lines allowed for comapnies in India. For example
If I am a company which has 1 incomming E1 line can I have SIP or IAX
extensions inside my office and receive that PSTN
One good step is to 'test' your public IPs against any mistake/hole like
this.
I've used http://www.ordb.org in the past for this purpose, others for
sure are available.
I would assume is a valuable feedback to provide to the folks from
[EMAIL PROTECTED], to have a more conservative
Keith Burns wrote:
I think you need to look at a few other factors.
...
2. Line power - Cisco uses one standard, other phones use another... but
Cisco is the 900# gorilla in the powered switch market... your call...
I'm curious about this point..
Most if not all vendors that support PoE are not
Christopher wrote:
Can anyone point me in a good direction for configuring SIP through a
PIX using 1:1 NAT. I have read anything I could get my hands on and
tried them all with very little success. I can get it to work through
the cheap little cable modem routers, but not this PIX.
I -can-
Matt Schulte wrote:
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What
we're trying to avoid is hardcoding the IP address in the ACL. We were
trying to match by TOS set by Asterisk however it seems we've run into a
snag where the packet TOS tends to get reset somewhere on our
Gonzalo,
Have you tried IAX, I see yo are behind NAT, and my experiences with IAX
behind NAT are much less painful :-)
I've FWD via IAX, receiveing calls (in fact, last time was a nearby
person in Portugal :-) that tested it).
One last thing, you mention dialup client, I guess she is not in
I would guess they really had some problems yesterday.
I had some failed calls in my * at home, had it rerouted via Nufone,
since the mother-in-law-retry timer was set too low, and I didn't want
to hear complaints when I arrived home :-). I'll try to switch it back
later today.
Ed Greenberg
Mike Diehl (Encrypted email preferred) wrote:
I'm trying to make an issue out of this because I think it needs to change and
I'm hoping people who are affiliated with these providers are reading this.
I was going to go with Packet8. I was going through the final checklist
before
Fabrício Zimmerer Murta wrote:
Oh, friend... I have realised just yesterday that's impossible to use
regular modems (say hayes/v90 33.6 or 56k) to plug asterisk to the world. I
can't figure out why. But they simply don't support it.
If you want to use your isdn modem to plug * to the world, it's
Jerry Glomph Black wrote:
I have a lot of experience, all of it pretty good, with various Sipura
products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones
connecting into Asterisk as clients. Good sound quality, great
reliability.
I've tried two of the units named in the
Remarks inline
Leonardo J. Tramontina wrote:
No, I'm not in USA!!
Besides this, my Asterisk is not making external calls; it is installed
for some tests...
I forgot to say... I'm making these tests in order to test the TE110P
card we bought.
- Original Message - From: Steven
Steve Kennedy wrote:
On Wed, Dec 01, 2004 at 02:53:50PM -0500, Kanuri, Seshu (Company IT) wrote:
Tell me which one can get me access to the LinkSys Linux using SSH? Does
Satori has this feature? I am not so concerned with Voice Shaping and
QOS at this time, but more interested in converting this
dean collins wrote:
Hi John,
I've been using Packet8 via a physical ATA and XP100 card for some time.
As far as I know it is not possible to connect to the Packet8 service
without the ATA.
If this is not the case I would be very interested to hear this.
In addition since moving to the USA I now
Brian Wilkins wrote:
You can only use g729 in pass-thru mode without paying for the licensing fees.
G729 is probably the best codec around. If you plan on having any sort of
thriving business based on VoIP, g729 would be the way to go. I don't suggest
PCMU or PCMA for production. The ATA will
Steve Underwood wrote:
Julio Arruda wrote:
Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the fax machine a la rxfax and the like.
I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine.
I know
not to use 729 for faxing. Which 'should' I use?
...
Hi
.
Julio Arruda wrote:
| Butit is quite weird they have such a small MTU. Many websites
| that have problems with Path MTU discovery would be broken by that
| (dumb websites, but still, way too many...).
The web sites in question aren't the problem. Again, it's things that
happen to the packets
Bastian Schern wrote:
Hi everybody,
I've got no success to get a friend in Bogota (Colombia) connected to my
Asterisk. He has got a ISDN Internet connection and the UDP packets will
be fragmented. It seems that the MTU of this connection is round about
400 to 500 Bytes. Therefore most UDP-SIP
Remarks inline:
Cian O'Sullivan wrote:
Hello,
I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on
their asterisk implementation. My local dealer says they are not
compatible with any open source implementations. Is there a phone
compatibility list somewhere?
First, 2
JOAO CARLOS MOURA wrote:
Thank you Michael,
I tried to use RFC2838 without success. Which another type?
Which endpoints (SIP Phones, ATA, ???) are we talking about ?
You need to match the configuration on the end-point, it may seem
obvious, but if you leave your IP phone doing dtmf inband, while
[EMAIL PROTECTED] wrote:
Do you have a list of those providers that use IAX?
http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers
is a good starting point..
Try a search on google, you would be surprised on how many of these will
pop...
-Original Message-
From: [EMAIL
Daniel Bichara wrote:
Hi Han,
Our company can offer you a SIP termination in Brazil up and running.
Daniel
IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio
de Janeiro.
Johannes van Hulst wrote:
Is there an up and running provider of SIP termination in Brazil?
I know
Steve Underwood wrote:
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
No, but if you find an E3 PCI card with nice Linux support there might
be people interested in helping to get it working with *.
Doesn't ImageStream have these (E3 and
Tim McKee wrote:
Guys:
I routinely run multiple phones over our satellite system (I'm the VP of
Network Services at SDN Global, a satellite bandwidth provider located in
Charlotte NC, US).
Just last week I went to West Palm Beach, FL US and turned up a 10 phone
emergency call center, complete
Sirs/Ladies,
Not sure if anyone saw anything like that before...
I was playing with an Asterisk setup with a Grandstream BT101 (1.0.5.11)
and www.voipjet.com (IAX2).
The other devices I have home (Sipura 3k and DTA-310) seem to work just
fine, but the Grandstream seems to suffer from one-way
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